Be aware with the user's manual. If you lose it somehow, or something like that, and you don't know the exact model of the device, it's difficult to find it out. There aren't any inscriptions about the model. The only "helpful" inscriptions are: "VoIP GSM Gateway" and "Made in Taiwan".

Dimension:

The device's demensions are:

* length: 14.5cm
* width: 17cm
* thickness: 4.1cm

The dimensions look for me good enough. The device is a little bigger, than the devices which I worked on before(ATAs, VoIP GSM phones...). Maybe Portech have to think about it?

All ports, connectors and indicators look stable and hopeful. The device is like a white bunker.

Accessory attachment:

First of all, lets say some words before continue to the attachment and configuration process. Leave the antenna installation on the last place. For now it is absolutely not necessary. I advise also, not to plug in the SIM card yet. Firstly lets attach the needed stuff for the basic configuration and after that we will plug in the antenna and the SIM card.

1) Connect the network cable to your Hub and to your device LAN port.2) Plug the adaptor in DC 12V socket and to the Power port on the device. The PWR light should turn red at the moment.

2. Configuration

Configuration via web page

The default IP address of the GSM Gateway is http://192.168.0.100. Prepare your network for this IP address. In Linux this can be done by setting the network device(eth0, eth1, eth2...) to IP address in this network range, for example 192.168.0.101. With "ifconfig" you can do that.

The first page after login is "System Information". There you can see model name, firmware version, codec version name.

In the left side you can click on the option you would like to set.

3. Route

3.1 Mobile to LAN

In this page you can set the routing rules which will transfer the calls incoming from Mobile to LAN. The maximum sets are 50.

When the GSM number of the MMv 370 GSM Gateway is called, this device transfers the call to URL according to the CallerID of the incoming call.

3.1.1 CID - CallerID, the numbers of incoming call

You could set the CID as the following formats:
1) The complete number, e.g. 02111111
2) The prefix part plus *, e.g. 021*. This format means any number starting with 021 will be accepted to transfer.
3) *, this means any incoming call is accepted to transfer.
4) N, this means the incoming call without showing its CID is accepted to transfer.

The priority of routing rules is that the CID with more digits gets the first priority.

Note: If the device is registered at Asterisk, you can enter any destination phone number. Also note that in the Asterisk, you need to set the route of destination phone numer.

4) Leave it blank or "N", this mean to refuse to transfer.
5) *, this means to transfer via 2-stage-dialing. The call will be answered with a prompt dial tone for the caller to press the IP address, proxy extension, or any phone number as destination. The caller press the IP address on the phone keys: 192*168*0*101# as 192.168.0.101.

3.1.3 Example of Mobile to LAN setting

1) Mobile to LAN: 0234*, 02111234
When the GSM numbers of the device is called, if the caller's prefix number are 0234, Mobile VoIP Gateway transfers the call to 02111234, then 02111234 rings.

Note: SIP extension or phone number needs to register at Asterisk. Also the phone number or Asterisk needs to set the route of destination phone number.

3.2 Mobile to LAN Speed Dial Settings

The call is answered with a prompt dial tone for the caller to press the "Num", and then the device connects the "URL" as destination.

Example: After you call the GSM number of the device and hear a dial tone, you press "0", then the lan phone of IP address: 192.168.0.117 rings.

3.3 LAN to Mobile Settings

In this page you can set the routing rules to transfer the calls incoming from LAN to Mobile. The maximum sets are 50.

When the LAN of the Mv 370 is called, the device transfers the call to Call Num according to the URL of the incoming call.

3.3.1 URL - the IP address or extension numbers of the incoming call.

The available formats are:

1) The complete IP address, e.g. 192.168.0.101
2) The extension numbers, e.g. 101
3) Part of an IP address plus "*", e.g. 192.168.0.*. This means the IP address starting with 192.168.0 would be accepted to transfer.
4) Part of the extension numbers plus, e.g. 10*. This means the extension numbers starting with 10 would be accepted to transfer.

3.3.2 Call Num - the phone numbers of destination.

The formats can be:

1) The complete number, e.g. 02111111
2) "*", this means to transfer via 2-stage-dialing. The call will be answered with a prompt dial tone for the caller to press the destination phone numbers, e.g. 02111111.
3) "#", this allow the caller with LAN phone dial directly the destination numbers.

Note: Mobile VoIP and incoming LAN phone have to be both registered at Asterisk. Asterisk has to have the routing rules to assign prefix of numbers to be transferred from Mobile VoIP. LAN to Mobile routing sets: *, #(You can dial on your LAN phone call any destination number with prefix "02". When your LAN phone and Mobile VoIP had registered and "09" prefix is setted the routing rules at Asterisk.)

4) #['d'n']['a'ppp], this means to do above routing, and to modify the numbers.

Note: 'd'n means to delete the number of prefix, 'a'ppp means to add 'ppp' prefix.

Example: #d3a02 means to call the registered numbers via one-stage-dialing. The numbers are modified to: delete 3 digits of prefix of the original numbers, then add 02 to be new prefix of the destination numbers.

4. Mobile

4.1 Mobile Status

In this page, you can get information of your GSM network and the latest operation.

Lets explain all fields:
1) Network registration - name of telecom carrier, which the SIM card of this device registers at.
2) SIM Card ID - SIM Card ID.
3) Signal Quality - place the antenna for higher signal, above 17 is better.
4) Incoming IP - IP address of the last incoming call from LAN.
5) Incoming IP Name - extension name of incoming call from LAN.
6) Outgoing IP - The IP address of the last outgoing call from LAN.
7) Incoming Mod - The CallerID of the last incoming call from Mobile.
8) Outgoing Mod - The destination numbers of the last outgoing call from Mobile.

4.2 Mobile Setting

In this page, you can change the values. For now, don't change these values, leave the default ones.

Note: You can change VoIP Volume and LAN DTMF Gain to fix the DTMF problem in Lan to Mobile operation. By changing the VoIP Gain and Mobile In Gain you can change the DTMF problem in Mobile to LAN operation.

5) CallerID - in Mobile to LAN operation, you can select "Clid" to display the incoming call numbers, or "Fix" to display fixed SIP user name on the destination phone.

6) Presentation CLIR - in LAN to Mobile operation, you select "Suppression" to hide the GSM numbers of the device, or "Invocation" to display it on the destination phone.

8) LAN Answer Mode - This is the LAN answer time while in LAN to Mobile routing.
Answered - when mobile side answers, then connects the call.
Alerted - when mobile side rings, then connects the call.
Income - when LAN side dials out, then connects the call soon.

9) Band Type - if your device is Quad band model, you need to choose your GSM frequency.

5. Network

In this section you can check the Network status, configure the WLAN Settings, LAN Settings and the SNTP Settings.

5.1 Network Status

Here you can see information of current Network.

5.2 Network Settings

You can check the current Network setting.

The default IP address is 192.168.0.100. You can change it to any available IP address, or select different IP type to suit your environment.

The settings are:
1) LAN Mode - select NAT
2) Fixed IP - the TCP/IP Configuration item is to setup the WAN port's network environment. You may refer to your current network environment to configure the system properly.
3) DHCP client - you can refer to your current network environment to configure the system properly.
4) PPPoE - if you have the PPPoE account from your Service Provider, please input the Username and the Password correctly.
5) After you input or modify the value, click the Submit button.

You can set the SIP and RTP port in this page. You have to contact to your Service Provider, to check what ports they are using.

6.3 Codec Settings

You can set the codec priority, RTP packet length. Please contact again with your ISP and follow their suggestions to setup this page. Don't forget to click on the "Submit" button.

6.4 Codec ID Setting

This page is for setting the Codec ID. Leave the default settings.

6.5 DTMF Setting

If your device has registered at Asterisk, select "2833", if not select "Inband DTMF".

6.6 RPort Setting

According to your Service Provider, what information is told to you, you can enable or disable the RPort.

6.7 Other Settings

According to your ISP information, you can setup the RFC and QoS. The Qo

QoS is using to set the voice packets' priority. If you set a value higher than 0, then the voice packets get the higher priority to the Internet. Note that, QoS function needs to cooperate with the others Internet devices.

7. NAT Trans

Here you can setup the STUN server. The STUN server is useful, if you are behind NAT. Again, check if you should have a STUN server with your Service Provider.

8. System Authority

Here you can change your login username and password.

Note: the defaults are:username: voippassword: 1234

9. Save Change

Don't forget this step everytime, when you submit any setting. After clicking "Save Change" then "Save", the system will restart and make the new changes operative.

10. Update

This section is for firmware upgrade and to restore the default settings.

Asterisk is the most popular and extensible open source telephone system in the world, offering flexibility, functionality and features not available in advanced, high cost proprietary business systems. Asterisk is a complete IP telephony platform for business, and can be downloaded for free.

Also on http://www.asteriskguru.com you can read our tutorials about the Asterisk installation, configuration, different types of hardphones, softphones, and much much more information.

You need:

Asterisk - the latest version in the 1.4 branch is 1.4.10. 1.2.24 is the latest in the 1.2 branch.

If you don't know how to install Asterisk, or you have some troubles with it, please read the following tutorial: Asterisk installation

Now lets configure the SIP account, that we will use. This one, which the Mv 370 - GSM Gateway will register itself to the Asterisk server.

sip.conf:

; GSM VOIP Gateway MV-370
[1006]
type=friend
secret=1006
qualify=yes ; keeps the NAT session open
nat=yes ; if you are behind NAT
host=dynamic
canreinvite=no ; set it to "no", just for insurance, if you are behind NAT
context=gsm ; the context in extensions.conf, which we will use for routing the calls

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Nedi (nedi at gmx dot ch)08 August 2009 16:57:25Hi
I have 2 MV-370 GSMGW and every 4-5 days my MV370 is not registered and I must reboot my MV-370 ..After reboot MV-370 device ist registered.
has anyone idea how to reboot the MV-370 every day with a script?
or how to reboot it wit a link ther is a rebootsys.cgi ??

best regards
nedi

mohamed (mw at aadc dot ae)11 May 2009 20:03:06please can you provide me with the availability of product voip gateway equipped with analog and gsm and has connection for one ip phone plus internet connection Ethernet , this is for home use between countries , i hope that you can feed back me soon,
in case you have this product we request one ip phone with connection to the internet directly and one of this products as sample then we will test then we will ask for quantities to be distributor or direct dealer. My phone no:- 00971505833466

Martin Candler (martin dot candler at aruncomputers dot co dot uk)09 October 2007 10:36:04Does this device pass the CID of the incoming Mobile call to asterisk or does it just pass the SIP Extension number of the GSM Gateway Device (due to the limitiations of the SIP Protocol)

Henk Oegema (henk at oegema dot com)08 October 2007 10:15:27I can't get DISA to work with the MV-370 and Asterisk 1.4.11
I have tried with both dtmfsetting=rfc2833 and dtmfsetting=inband. (also in my sip.conf I cahnged it accordingly to the settings in the MV370).

In the MV-370 user manual (at page 42) is written:
dtmfmode=inband;Very important for DISA to work. However this doesn't work (for me) either.

Any other settings which are of importance?

Grds. Henk Oegema

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